Among conventional audio signal coding methods and decoding methods, an international standard scheme of the ISO/IEC, commonly called MPEG (Moving Picture Experts Group) scheme, is well known. Currently, ISO/IEC 14496-3, commonly called MPEG-4 GA (General Audio Coding), is one of the coding schemes that are widely applicable and achieve high quality sound even at low bit rates (see Non-Patent Reference 1). There are many extended specifications of this scheme that are currently being standardized.
Among them is a low-delay technique for reducing a delay that occurs in coding and decoding. An example is the Low Delay AAC (Advanced Audio Coding) scheme defined by MPEG-4 Audio (ISO/IEC 14496-3) which is an ISO/IEC international standard. Other examples include techniques disclosed by Patent Reference 1 and Non-Patent Reference 2.
Hereinafter, a conventional audio signal coding method and decoding method of Non-Patent Reference 2 shall be described.
FIG. 1 is a configuration diagram of a conventional audio signal coding apparatus. An audio signal coding apparatus 100 in the figure is an apparatus characterized particularly in reducing a delay in its processing. The audio signal coding apparatus 100 includes an auditory redundancy eliminating unit 101 and an information redundancy eliminating unit 102.
The auditory redundancy eliminating unit 101 eliminates auditory redundancy from an input audio signal. More specifically, it eliminates components that are inaudible by humans from the audio signal based on aural characteristics of humans. The auditory redundancy eliminating unit 101 includes an auditory model 103, a pre-filtering unit 104, and a quantizing unit 105.
The auditory model 103 is an important element for determining the audio artifact of coded audio signals. It screens the sounds and levels of frequency components inaudible by humans, by using a technique well known to those skilled in the art, such as temporal masking or simultaneous masking. As a result, it adaptively calculates the level, in each frequency band, of the sounds of frequency components audible by humans, for input audio signals. The auditory model 103 outputs to the pre-filtering unit 104 information indicating, based on the calculation result, what kind of filter the pre-filtering unit 104 should use. Meanwhile, the auditory model 103 outputs the information after including it in a coded sequence of an audio signal which is an output signal of the audio signal coding apparatus. The auditory model 103 is for example an auditory model described in the specification of the MPEG-1 Layer III (commonly called MP3). An input digital audio signal sequence is first inputted to the auditory model 103.
Based on the information provided by the auditory model 103 indicating what kind of filter should be used, that is, based on a value indicating the level, in each band, of frequency components audible by humans, the pre-filtering unit 104 eliminates with a filter the sounds of the components at the level inaudible by humans from the input digital audio signal sequence. By doing so, the pre-filtering unit 104 outputs an audio signal sequence with no inaudible components. The pre-filtering unit 104 is structured with plural linear prediction filters as disclosed in Non-Patent Reference 2.
The quantizing unit 105 quantizes the audio signal sequence received from the pre-filtering unit 104 by rounding off values less than an integer, and outputs an audio signal sequence which is an integer.
In such a manner, the auditory redundancy eliminating unit 101 eliminates, from input audio signal sequences, components inaudible by humans, and outputs audio signal sequences that are quantized into an integer.
The information redundancy eliminating unit 102 eliminates redundant information from the audio signal sequences received from the auditory redundancy eliminating unit 101 so as to enhance the coding efficiency. The information redundancy eliminating unit 102 includes a lossless coding unit 106.
The lossless coding unit 106 has conventionally been proposed, and employs a method such as Huffman coding, a technique well known by those skilled in the art. The audio signal sequences inputted to the lossless coding unit 106 are previously quantized into integers by the above mentioned quantizing unit 105. So, the lossless coding unit 106 which performs Huffman coding, for example, eliminates redundant information from the integers so as to enhance the coding efficiency.
With the above structure, the conventional audio signal coding apparatus 100 outputs both of the following as a coded sequence: information indicating what kind of prefilter was used by the pre-filtering unit 104, that is, information indicating the linear prediction coefficients that structure the pre-filtering unit 104; and an audio signal sequence (information) coded by the lossless coding unit 106.
Next, a conventional audio signal decoding apparatus shall be described.
FIG. 2 is a configuration diagram of a conventional audio signal decoding apparatus. An audio signal decoding apparatus 200 in the figure decodes an audio signal which has been coded. The audio signal decoding apparatus 200 includes a lossless decoding unit 201 and a post-filtering unit 202.
The lossless decoding unit 201 decodes an audio signal sequence by performing lossless-decoding on a coded sequence outputted from the lossless coding unit 106.
The post-filtering unit 202 structures a postfilter (inverse of the filter used by the pre-filtering unit 104) from a decoded, linear prediction coefficient sequence. The post-filtering unit 202 post-filters the audio signal sequence which has been lossless-decoded by the lossless decoding unit 201, to eventually output an audio signal sequence obtained through the post-filtering.
By using the audio signal coding apparatus of FIG. 1 and the audio signal decoding apparatus of FIG. 2 in the above manner, the delay is made smaller than in the case of using the coding and decoding methods such as AAC. This is because there is no longer a delay for a batch orthogonal transformation process in which one frame of the scheme such as AAC has 1024 samples, for example, and the delay from the pre-filtering and post-filtering is small. As a consequence, a low delay can be achieved.    Patent Reference 1: International Patent Application Publication No. WO2005/078705    Non-Patent Reference 1: ISO/IEC 14496-3: 2005 “General Audio Coding”    Non-Patent Reference 2: Conference Paper “Perceptual Audio Coding Using Adaptive Pre- and Post-Filters and Lossless Compression” (IEEE Transaction on Speech and Audio Processing, vol. 10, No. 6, September 2002)